The proliferation of data transport networks, most notably the Internet, is causing a revolution in telephony and other forms of real-time communication. Businesses that have been accustomed to having telephony traffic and data traffic separately supported over different systems and networks are now moving towards so-called “converged networks” wherein telephone voice traffic and other forms of real-time media are converted into digital form and carried by a packet data network along with other forms of data. Now that the technologies are feasible to support it, voice over data transport offers many advantages in terms of reduced capital and operating costs, resource efficiency and flexibility.
For example, at commercial installations, customer premise equipment investments are substantially reduced as most of the enhanced functions, such as PBX and automatic call distribution functions, may reside in a service provider's network. Various types of gateways allow for sessions to be established even among diverse systems such as IP phones, conventional analog phones and PBXs as well as with networked desktop computers.
To meet the demand for voice over data transport, service providers and network equipment vendors are faced with the challenges of establishing new protocols and standards, recognizing new business models, implementing new services, and designing new equipment in a way that would have been difficult to imagine twenty years ago.
For establishing a communications session in a network, new protocols and control architectures have emerged. It is worth noting that these have been inspired by the migration to a voice over data but are not necessarily limited to such an environment. In some respects the protocols and control architectures described next may be used to establish calls through any form of transport.
Both the ITU H.323 standard and the IETF's Session Initiation Protocol (SIP) are examples of protocols which may be used for establishing a communications session among terminals connected to a network. The SIP protocol is described in IETF document RFC 2543 and its successors. Various architectures have been proposed in conjunction with these protocols with a common theme of having an address resolution function, referred to as a “location server,” somewhere in the network to maintain current information on how to reach any destination and to control features on behalf of users.
In a SIP-controlled network, a variety of features and services may be implemented via a SIP server. These features include Centrex-type calling features (call forwarding) as well as advanced features such as SIP presence support, location management, and Find-Me capability. A typical feature in a communications network is call blocking. Call blocking is defined as the ability for an administrator to place outgoing call restrictions on individual users. For instance, administrators may make restrictions on outgoing calls to certain international phone number ranges for different individual users.
In addition to regular outgoing calls that a user directly places, outgoing calls can also be initiated as a side result of a feature invocation. For instance, call forwarding, call transfer, and other features, can result in an outgoing call indirectly. However, administrators may wish to apply different policies to these type of indirect calls, as opposed to direct calls. For instance, it might be ok to dial a Long Distance call directly from a business location for a particular user. But, it might not be acceptable for that same user to be able to forward calls to Long Distance, as this may lead to a fraudulent use of the phone from outside the business location.
This situation is depicted in FIG. 3A. FIG. 3A shows three parties which have specific permissions to place calls among one another in the context of a given communications system (not explicitly shown). In particular, Party A 301 may make calls to Party B 303, and Party B 303 may make calls to Party C 305. However, the network serving these parties is configured to prevent Party A from placing calls directly to Party C. For example, there may be long distance charges or tolls incurred in calling Party C. Party B may be authorized to incur such long distance charges and is allowed to place calls to Party C. On the other hand, call attempts from Party A to Party C may be blocked to avoid costly calls by Party A. A good example might be in a corporate setting wherein Party B is an employee of a corporation and is allowed to reach remote Party C, even if the call involves long distance charges. Party A may correspond to a courtesy telephone placed in the lobby at a business location or may be assigned to a contractor or temporary employee within the company. Party A is allowed to make calls within the company, but not to make outbound long distance calls.
As shown in FIG. 3B, a problem arises when Party B activates a call forwarding feature, or any other feature that triggers outbound calls. A call from Party A may be forwarded or redirected to Party C, resulting in a call that would have normally been blocked otherwise. Aside from providing a possible mechanism for fraud or for inadvertent accumulation of charges, this circumvention of normal screening may be a problem for other reasons. The unwelcome call from User A to the unaware User C might, for example, compromise personal security (harassment, stalking, annoyance calls), privacy or confidentiality (attorney/client or doctor/patient), or the security of a facility.
Without placing undue restrictions on Party B, it is desirable to control the ability of Party A to cause calls to Party C, regardless of what routing features are invoked by Party B.